The PlayoutDelayOracle was responsible for making sure the PlayoutDelay header extension was successfully propagated to the receiving side. Once it was determined that the receiver had received a frame with the new delay tag, it's no longer necessary to propagate. The issue with this implementation is that it is based on max extended sequence number reported via RTCP, which makes it often slow to react, could theoretically fail to produce desired outcome (max received > X does not guarantee X was fully received and decoded), and added a lot of code complexity. The guarantee of delivery can in fact be accomplished more reliably and with less code by making sure to tag each frame until an undiscardable frame is sent. This allows containing the logic fully within RTPSenderVideo. Bug: webrtc:11340 Change-Id: I2d1d2b6b67f4f07b8b33336f8fcfcde724243eef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168221 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30473}
107 lines
3.1 KiB
C++
107 lines
3.1 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef COMMON_TYPES_H_
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#define COMMON_TYPES_H_
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#include <stddef.h> // For size_t
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#include <cstdint>
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namespace webrtc {
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struct FrameCounts {
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FrameCounts() : key_frames(0), delta_frames(0) {}
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int key_frames;
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int delta_frames;
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};
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// Callback, used to notify an observer whenever frame counts have been updated.
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class FrameCountObserver {
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public:
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virtual ~FrameCountObserver() {}
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virtual void FrameCountUpdated(const FrameCounts& frame_counts,
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uint32_t ssrc) = 0;
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};
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// Callback, used to notify an observer when the overhead per packet
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// has changed.
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class OverheadObserver {
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public:
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virtual ~OverheadObserver() = default;
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virtual void OnOverheadChanged(size_t overhead_bytes_per_packet) = 0;
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};
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// ==================================================================
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// Video specific types
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// ==================================================================
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// TODO(magjed): Move this and other H264 related classes out to their own file.
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namespace H264 {
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enum Profile {
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kProfileConstrainedBaseline,
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kProfileBaseline,
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kProfileMain,
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kProfileConstrainedHigh,
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kProfileHigh,
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};
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} // namespace H264
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struct SpatialLayer {
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bool operator==(const SpatialLayer& other) const;
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bool operator!=(const SpatialLayer& other) const { return !(*this == other); }
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unsigned short width;
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unsigned short height;
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float maxFramerate; // fps.
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unsigned char numberOfTemporalLayers;
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unsigned int maxBitrate; // kilobits/sec.
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unsigned int targetBitrate; // kilobits/sec.
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unsigned int minBitrate; // kilobits/sec.
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unsigned int qpMax; // minimum quality
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bool active; // encoded and sent.
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};
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// Simulcast is when the same stream is encoded multiple times with different
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// settings such as resolution.
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typedef SpatialLayer SimulcastStream;
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// Minimum and maximum playout delay values from capture to render.
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// These are best effort values.
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//
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// A value < 0 indicates no change from previous valid value.
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//
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// min = max = 0 indicates that the receiver should try and render
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// frame as soon as possible.
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//
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// min = x, max = y indicates that the receiver is free to adapt
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// in the range (x, y) based on network jitter.
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//
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// Note: Given that this gets embedded in a union, it is up-to the owner to
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// initialize these values.
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struct PlayoutDelay {
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PlayoutDelay(int min_ms, int max_ms) : min_ms(min_ms), max_ms(max_ms) {}
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int min_ms;
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int max_ms;
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static PlayoutDelay Noop() { return PlayoutDelay(-1, -1); }
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bool IsNoop() const { return min_ms == -1 && max_ms == -1; }
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bool operator==(const PlayoutDelay& rhs) const {
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return min_ms == rhs.min_ms && max_ms == rhs.max_ms;
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}
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};
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} // namespace webrtc
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#endif // COMMON_TYPES_H_
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